It is well known that the transmission of data in digital form provides for increased signal to noise ratios and increased information capacity along the transmission channel. There is however a continuing desire to further increase channel capacity by compressing digital signals to an ever greater extent. In relation to audio signals, two basic compression principles are conventionally applied. The first of these involves removing the statistical or deterministic redundancies in the source signal whilst the second involves suppressing or eliminating from the source signal elements with are redundant insofar as human perception is concerned. Recently, the latter principle has become predominant in high quality audio applications and typically involves the separation of an audio signal into its frequency components (sometimes called “sub-bands”), each of which is analysed and quantised with a quantisation accuracy determined to remove data irrelevancy (to the listener). The ISO (International Standards Organisation) MPEG (Moving Pictures Expert Group) audio coding standard and other audio coding standards employ and further define this principle. However, MPEG (and other standards) also employs a technique know as “adaptive prediction” to produce a further reduction in data rate.
The operation of an encoder according to the new MPEG-2 AAC standard is described in detail in the draft International standard document ISO/IEC DIS 13818-7. This new MPEG-2 standard employs backward linear prediction with 672 of 1024 frequency components. It is envisaged that the new MPEG-4 standard will have similar requirements. However, such a large number of frequency components results in a large computational overhead due to the complexity of the prediction algorithm and also requires the availability of large amounts of memory to store the calculated and intermediate coefficients. It is well known that when backward adaptive predictors of this type are used in the frequency domain, it is difficult to further reduce the computational loads and memory requirements. This is because the number of predictors is so large in the frequency domain that even a very simple adaptive algorithm still results in large computational complexity and memory requirements. Whilst it is known to avoid this problem by using forward adaptive predictors which are updated in the encoder and transmitted to the decoder, the use of forward adaptive predictors in the frequency domain inevitably results in a large amount of “side” information because the number of predictors is so large.
It is an object to the present invention to overcome or at least mitigate the disadvantages of known prediction methods.
This and other objects are achieved by coding an audio signal using error signals to remove redundancy in each of a plurality of frequency sub-bands of the audio signal and in addition generating long term prediction coefficients in the time domain which enable a current frame of the audio signal to be predicted from one or more previous frames.